[asterisk-users] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work

[Gareth Blades]

On 05/12/14 16:46, Olli Heiskanen wrote:
> INVITE that Asterisk (at port 5070) receives:
> PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046
> INVITE sip:660 at testers.com 
> <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0
> Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177>
> Via: SIP/2.0/UDP 
> PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0
> Via: SIP/2.0/UDP 
> AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z-
> Max-Forwards: 69
> Contact: <sip:771 at AST.ER.ISK.IP:38699;transport=UDP>
> To: <sip:660 at testers.com <mailto:sip%3A660 at testers.com>;transport=UDP>
> From: "771"<sip:771 at testers.com 
> <mailto:sip%3A771 at testers.com>;transport=UDP>;tag=41030177
> Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, 
> INFO, SUBSCRIBE
> Content-Type: application/sdp
> Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
> User-Agent: Z 3.2.21357 r21367
> Allow-Events: presence, kpml
> Content-Length: 239
>
> v=0
> o=Z 0 0 IN IP4 AST.ER.ISK.IP
> s=Z
> c=IN IP4 AST.ER.ISK.IP
> t=0 0
> m=audio 8000 RTP/AVP 3 110 8 0 98 101
> a=rtpmap:110 speex/8000
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv

This client is saying it only supports speex and iLBC and would prefer 
them in that order.
Your sip.conf appears to only permit alaw, ulaw and gsm so there is no 
mutual supported codec and hence the call fails.

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